Vodia PBX

Maybe we should vote - should we have just one SMTP server setting for the whole system?

No, the statistics is kept inside each queue.

G.722 isn't G.722... Polycom also uses G.722.1 ©, which is not compatible with the good old G.722 that the PBX supports. Maybe there is a problem (what is in the SDP)?

At the same time? Interesting idea, but currently that is not possible.

What has changed is that the 0 is only respected if it is recording already.

We have one in our lab. Essentially it is a standard PC embedded in the 19" rack. It runs well with the pbxnsip PBX. IMHO PRO is that it is a rock solid, 1U 19" appliance that makes a pretty good impression. The PSTN termination is just great. It is something that you want to see in your rack if you hate trouble and love superior audio quality. The CON is that the PC is relatively weak (reportedly around 50 parallel calls if nothing special is going on) compared to latest stand-alone servers.

We took it only out during the recording phase. It is still available during the annoucement and after the recording (if the user presses #).

No, DTMF sensititity is "hard coded". We did a major upgrade maybe around a year ago, and so far it was reportedly okay. If situations with heavy jitter I can imagine there will be trouble. Of course, if you can you should always use out of band and push everyone to support out of band. Maybe you can get a Wireshark file so we can extract the audio for reproducing the problem in the lab.

Hmm. We need to verify that...

Strange. Well, usually any call should be cleared after two hours, unless yo uchange the setting on system level. What happens if you hit the delete button for that link?

The CDR are sent from the domain. Did you set up the domain email address? It should be right next to the try button. Otherwise, try setting the clock close to midnight and turn the logging for email things on and see what happens.

As far as RFC3325 says, that is IMHO not correct. In previous versions of the PBX was was a little bit "messy". The idea of RFC3325 is that the From header can go through all proxies unchanged, and the authentication part is done on other headers. I know that the support out there for this is weak. Probably the biggest problem is that many ITSP are using SER, which did not support this for a long time. But somehow we have to get started with the standard, that is why the latest and the greatest is strictly according to the RFC.

I would be only concerned if you are running latest release...

Good catch! Lets see if it works in the next.

STUN is not supported any more. It just causes too many problems with firewalls and is a support and stability nightmare. It is simply not a clean solution. Having a trunk working 95 % is not an option for a PBX. If you want to get a PBX working behind NAT you must manually set up the network - see for example http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses.

Well, there is a new release 2.1.3 image that should avoid the problem.

ringtones.xml (I attached the latest version). ringtones.xml

Well, you need to edit that file above and put it into the html directory... This requires a service restart.

Those files were collected by the garbage collector inside the PBX - should not happen but if it happens, it writes a log so that later we can find out why. You can just remove these files.

Ouch.

Okay, I assume that 40@buero-newyork.com is a extension? And I assume that sippwd is the web password for that extension? If that does not work, try including the parameter "auth" in the URL that contains the base64-encoded just in the CURL style user:password (like auth=40@buero-newyork.com:sippwd). Maybe there is something wrong with the CURL way of answering challenges.

We have made version 2.1.3 available. It has some important bug fixes and also comes with a handful of interesting new features. Please see http://wiki.pbxnsip.com/index.php/Release_Notes_2.1.3 for the release notes and http://www.pbxnsip.com/software for the download links.

That makes sense to me - the gateway should present the +12225555555 to the caller and use the 269@10.1.1.254 for authentication.

By default, the affinity mask is always "1". Seems reasonable to me and will be only a problem when there is something else critical running on that system. So usually you don't have to worry about this setting. See also http://wiki.pbxnsip.com/index.php/Processor_Affinity.

Does it work if you do click2dial from the web interface? I guess "hhttp" is a typo? I guess you saw http://wiki.pbxnsip.com/index.php/Click_To_Dial.